JP2020510240A - Real-time sound processor - Google Patents

Abstract

The present disclosure includes an acoustic processing network that includes a digital signal processor (DSP) operating at a first frequency and a real-time acoustic processor (RAP) operating at a second frequency that is higher than the first frequency. The DSP receives a noise signal from at least one microphone. The DSP generates a noise filter based on this noise signal. The RAP receives the noise signal from the microphone and the noise filter from the DSP. The RAP then generates an anti-noise signal used for active noise cancellation (ANC) based on the noise signal and the noise filter.

Description

BACKGROUND Active noise cancellation (ANC) can be used to reduce the amount of environmental noise heard by a user wearing headphones. In the ANC, a noise signal is measured and a corresponding anti-noise signal is generated. An anti-noise signal is an inverse signal that approximates a noise signal. The noise signal and the anti-noise signal destructively interfere to remove some or all of the environmental noise from the user's ear. To generate an accurate anti-noise signal for high quality ANC, the corresponding system needs to react quickly to changes in environmental noise. If not responded quickly, delays are undesirable for ANC because noise cannot be properly canceled. If the correction circuit does not react quickly, erroneous amplification of noise, burst of anti-noise that does not cancel the noise signal, and the like may be caused. Further, when listening to music on headphones, ANC can be even more complicated. In some cases, the ANC may not be able to distinguish between noise and low frequency music. As a result, the music signal is erroneously removed together with the noise signal.

  Aspects, features and advantages of embodiments of the present disclosure will become apparent from the following description of embodiments with reference to the accompanying drawings.

1 is a schematic diagram illustrating an exemplary sound processing network. FIG. 2 is a schematic diagram illustrating an input / output (I / O) of an exemplary real-time sound processor (RAP). FIG. 2 is a schematic diagram illustrating an exemplary sound processing network for sharing compressor states. FIG. 1 is a schematic diagram illustrating an exemplary sound processing network for performing audio input equalization. FIG. 2 is a schematic diagram illustrating an example RAP architecture. FIG. 4 is a schematic diagram illustrating another exemplary RAP architecture. FIG. 4 is a schematic diagram illustrating an example programmable topology of a RAP. FIG. 4 is a schematic diagram illustrating another example programmable topology of a RAP. FIG. 3 is a schematic diagram illustrating a biquad filter structure. 5 is a flowchart illustrating an exemplary method of operating a sound processing network.

DETAILED DESCRIPTION This specification discloses an exemplary sound processing network. The network includes a digital signal processor (DSP) operating at a first frequency and a RAP operating at a higher second frequency. DSPs can generate powerful noise filters to support the generation of accurate anti-noise signals. The DSP transfers and implements these noise filters to the RAP. RAPs operate faster than DSPs and can respond quickly to changes in hearing. As a result, the delay can be reduced, and a precise anti-noise signal can be maintained. The filters provided by the DSP may depend on user input and / or environmental changes. For example, when a user moves from a quiet environment to a noisy environment, the DSP may change the noise filter. As another example, a RAP can use a compressor circuit to control an adjustable amplifier in a pair of headphones. The compressor circuit can adjust the amplifier based on the compressor state. This may limit the rate of change of the volume of the anti-noise signal. Unless rapid changes in volume are limited, clipping of signals that the user perceives as pops or clicks may occur. The DSP can cope with such a change in volume by adjusting the compressor state of the RAP based on the change in the environmental sound. Also, DSPs and RAPs can support environment awareness when receiving input from a user. The environment recognition may be associated with a predetermined frequency band, for example, a frequency band related to human speech. The DSP can generate a noise filter that increases the gain of a noise signal in a predetermined frequency band. Accordingly, the RAP amplifies the associated band when generating the anti-noise signal. As a result, it is possible to emphasize sound (for example, speech) in a corresponding frequency band and cancel environmental noise. The DSP can also provide a tuned audio signal based on the audio signal and the expected frequency response of the audio processing network. The RAP can use this conditioned audio signal as a reference point when performing ANC. This allows the RAP to drive the entire output to the expected audio output instead of driving the output to zero and canceling a portion of the audio signal (eg, low frequency music). RAP is also designed to transfer anti-noise signals to one or more class G controllers for controlling the class G amplifier of a digital-to-analog converter (DAC) in headphones. Thus, gain control of the anti-noise signal is supported, and signal disturbance is further reduced. In addition, RAP can implement various noise filters from the DSP using biquad filters. The biquad filter filters the signal samples naturally when storing them, which can cause a partial loss of signal fidelity. As an example, RAP uses an implemented biquad filter to amplify the sample, then quantize the sample, and then attenuate the sample. Processing in this order reduces and therefore minimizes quantization errors. Therefore, a more accurate anti-noise signal can be obtained.

  FIG. 1 is a schematic diagram illustrating an exemplary sound processing network 100 that may be used for ANC. The sound processing network 100 includes a DSP 110 operating at a first frequency and a RAP 120 operating at a second frequency higher than the first frequency, where the second frequency is higher than the first frequency. For example, the DSP 110 can operate at 96 kilohertz (kHz) or less. In many cases, DSP 110 may operate at about 48 kHz (eg, the first frequency). RAP 120 can operate at frequencies up to about 6.144 megahertz (MHz). As a specific example, RAP 120 may operate at 0.768 MHz, 1.5 MHz, 3 MHz, and / or 6.144 MHz (eg, a second frequency). DSP 110 may be highly programmable and may have significant processing power. However, the RAP 120 operates at a higher frequency and thus can operate much faster than the DSP 110. Thus, RAP 120 responds with a much lower delay than DSP 110. Therefore, the sound processing network 100 generates a sound filter using the DSP 110 and controls the network 100. On the other hand, the RAP 120 responds quickly to environmental changes using the audio filters provided by the DSP 110 when performing ANC and similar functions.

  DSP 110 is any special processing circuit optimized for processing digital signals. DSP 110 supports many different functions. For example, the sound processing network 100 can operate with a set of headphones. When playing music or other audio to a user, DSP 110 may receive audio input in digital form from memory and / or a general purpose processing unit. The DSP 110 can generate an audio signal 143 corresponding to the audio input. The audio signal 143 is a stream of digital data including audio reproduced to the user via the speaker 136. For example, the DSP 110 can generate a left audio signal 143 applied to the left ear of the user and a right audio signal 143 applied to the left ear of the user. As described below, in some examples, DSP 110 may generate a pair of audio signals 143 for each ear. In addition, the DSP 110 generates various noise filters applied to the audio signal 143, for example, to compensate for noise generated by the operation of the audio processing network 100.

  Further, when providing the ANC, the DSP 110 can generate a noise filter used when generating an anti-noise signal. In this case, DSP 110 receives one or more noise signals 144 from one or more microphones 137. Microphone 137 may include a feedforward (FF) microphone located outside the user's ear canal. The FF microphone is arranged to record environmental noise before the user experiences it. Accordingly, the DSP 110 can use the noise signal 144 from the FF microphone 137 to determine the noise that the user is expected to experience in the near future. Next, the DSP 110 can generate a noise filter based on the noise signal 144. This noise filter can then be used (eg, by RAP 120) to cancel noise signal 144 by generating an anti-noise signal. Microphone 137 may also include a feedback (FB) microphone. The FB microphone is located inside the user's ear canal. Therefore, the FB microphone 137 is arranged to record the noise that the user is actually experiencing after applying the anti-noise signal. Therefore, the signal error can be corrected by repeatedly adjusting the noise filter of the anti-noise signal using the noise signal 144 from the FB microphone 137. Note that the use of at least the FF microphone and the FB microphone 137 (for example, four or more microphones 137) for each ear can achieve the best performance. However, ANC can be realized using only the FF microphone or the FB microphone 137.

  DSP 110 can communicate with RAP 110 by providing control and configuration parameters 141. The parameters 141 may include a noise filter for generating an anti-noise signal, a noise filter for adjusting the audio signal 143, and commands for performing various functions. The RAP 110 receives a noise filter from the DSP 110 via the control and setting parameters 141 and can perform various audio processing. RAP 110 may be any digital processor optimized for low-latency digital filtering. Also, the RAP 120 can receive the noise signal 144 from the microphone 137 when performing ANC. RAP 120 may generate an anti-noise signal based on noise signal 144 and a noise filter from DSP 110. The anti-noise signal is transferred to the speaker 136 and used for ANC. Further, the RAP 120 can change the audio signal 143 output to the speaker 136 using the noise filter from the DPS 110. Accordingly, the RAP 120 can mix the anti-noise signal and the modified audio signal 143 to generate the output signal 145. Thereafter, the output signal 145 is transferred to the speaker 136 and reproduced by the user. Speaker 136 may be any headphone speaker. In some cases, microphone 137 may be physically fixed to a pair of speakers 136 (eg, a left headphone speaker and a right headphone speaker).

  As described above, since the RAP 120 can operate at a higher frequency than the DSP 110, it can operate with a lower delay than the DSP 110. For example, the DSP 110 can generate a noise filter based on a change in the level of normal noise in the user's surrounding environment. For example, as the user moves from a noisy room to a quiet room, DSP 110 may generate different noise filters. Because such changes are relatively slow, the delay of the DSP 110 is sufficient to accommodate such changes. On the other hand, the RAP 120 responds quickly to specific noise changes by applying a noise filter. For example, the RAP 120 can reduce a specific sensory noise such as a falling sound of a board, a crying sound of a child, or a closing sound of a door by generating an anti-noise signal using a noise filter corresponding to a noisy room. . As a specific example, the delay between receiving a sample of the noise signal 144 from the microphone 137 and transferring the corresponding sample of the anti-noise signal to the speaker 136 is less than about 100 microseconds (eg, about 5 microseconds). There may be.

  Also, the DSP 110 may be configured to obtain and process various RAP states 142 from the RAP 120. RAP state 142 may include various states and other intermediate signals used by RAP 120's finite state machine. The DSP 110 can use the RAP state 142 to determine control and configuration parameters 141. Accordingly, the RAP state 142 allows the DSP 110 to dynamically control the RAP 120 by providing feedback from the RAP 120 to the DSP 110. For example, as described below, RAP 120 can utilize audio compression and RAP state 142 can include a compression state. This allows the DSP 110 to dynamically change the compression performed on the RAP 120. Note that the RAP 120 can use the pause to display important events on the DSP 110, such as signal clipping, completion of feathering, instability detected on the left channel, and instability detected on the right channel. it can. Such pauses can be enabled / disabled, respectively, using programmable registers.

  As shown in FIG. 1, DSP 110 and RAP 120 operate at different frequencies in the digital domain, while speaker 136 and microphone 137 operate in the analog domain. The sound processing network 100 utilizes various components to support conversion between domains and frequency velocities. For example, interpolator 135 may be used to increase the frequency of audio signal 143 from a first frequency used by DSP 110 to a second frequency used by RAP 120. Interpolator 135 is any signal processing component that uses interpolation to increase the frequency of the signal by increasing the effective sampling rate. The audio signal 143 may be sampled at a rate audible to the human ear. The interpolator 135 can increase the sampling rate of the audio signal 143 input to the RAP 120 (for example, from 48 kHz to 384 kHz). Thus, the interpolated audio signal 143 may be considered oversampled for audio reproduction. In other words, the bandwidth associated with the auditory signal is about 20 kHz. According to the Nyquist criterion, 40 kHz sampling is required to fully capture a 20 kHz signal. Therefore, the audio signal 143 of the RAP 120 may be considered to be strongly oversampled.

  Communication between the RAP 120 and the DSP 110 (along the noise signal path) can occur via a decimator 134. Decimator 134 is any signal processing component that utilizes decimation to reduce the frequency of the signal by reducing the effective sampling rate. Accordingly, the decimator 142 can be utilized to reduce the frequency of the signals (eg, the RAP status signal 142 and the noise signal) from the second frequency used by the RAP 120 to the first frequency used by the DSP 120. . In other words, interpolator 135 upconverts / upsamples the signal, and decimator 134 downconverts / downsamples the signal.

  The network 100 also uses one or more digital-to-analog converters (DACs) 131 and one or more analog-to-digital converters (ADCs) 133 to convert between the analog domain and the digital domain. The DAC 131 is a signal processing component that converts a digital signal into an analog signal. The ADC 33 is a signal processing component that converts an analog signal into a digital signal. Specifically, ADC 133 receives analog noise signal 144 from microphone 137 and converts this signal to the digital domain for use by RAP 120 and DSP 110. On the other hand, the DAC 131 receives the digital output signal 145 (including the anti-noise signal and / or the audio signal 143) from the RAP 120 and converts the output signal 145 into an analog format that can be output by the speaker 136. In some examples, a modulator 132, eg, a delta-sigma modulator, may be utilized to support the DAC 131. The modulator 132 is a signal component that reduces the number of bits of the digital signal and increases the frequency as a preprocessing step before the digital / analog conversion by the DAC 131. Modulator 132 may support DAC 131, but may not be used in some examples. Note that the modulator 132 and the DAC 131 may have a fixed transfer function. Thus, the RAP 120 may be the last block in the audio processing chain with high configurability.

  The DAC 131 can increase the volume of the output signal 143 to a level suitable for reproduction by the speaker 136 using an amplifier such as a class G amplifier. The network 100 can control the amplification of the DAC 131 using an amplifier controller 130 such as a class G amplifier controller. For example, a low volume output signal 145 (eg, an anti-noise signal for quiet environments and / or silence of the audio signal 143) requires little amplification. Conversely, a loud output signal 145 may require a large amplification (eg, a large noise and / or a large anti-noise signal for the loud music of the audio signal 143). Because the anti-noise signal output from DAC 131 is potentially very variable, sudden changes in volume may occur. Such sudden changes can cause speech disturbance. For example, if the output signal 145 suddenly increases beyond the capabilities of the DAC 131 amplifier, the sudden change from silence to a loud anti-noise signal (e.g., sudden applause in a quiet room) will cause the signal to be amplified by the DAC 131 amplifier. May cause clipping. Such clipping is experienced by the user as a pop or click. To avoid such disturbances, RAP 120 forwards a copy of the anti-noise signal to digital amplifier controller 130. Amplifier controller 130 can support adjustment of DAC 131 amplifier based on the level of the anti-noise signal (eg, by changing the applied voltage). The amplifier controller 130 can predict a potential change of the output signal 145 by dynamically considering a change in the anti-noise signal. Next, the amplifier controller 130 can change the amplification setting of the DAC 131 based on the change in the anti-noise signal (and / or the change in the audio signal 143) to save power by reducing the amplification, Alternatively, clipping can be prevented by increasing the amplification. The functions generally described above with reference to FIG. 1 are described in more detail below. Note that each of these functions can be enabled alone or in combination based on user input (eg, ANC can be enabled with or without voice input).

  Note that the noise entering the user's ear depends on many factors, including the shape of the head and ears and the tightness and fit of the headphones. Also, the audio signal generated by the headphones may depend on the tightness between the user's ears and the headphones. In other words, the transfer function of the headphones may depend on the tightness. Due to these variability, the design of a single ANC filter that produces an anti-noise signal may not be optimal for all users. Adaptive ANC results in an ANC filter design that is optimized for the current user. Since the DSP 110 can use the noise signal 144 of the FF and the FB microphone 137, an adaptive ANC can be realized. DSP 110 may estimate the transfer function between FF and FB noise signal 144 for a particular user during the calibration phase. For example, the DSP 110 can determine the noise in the ear based on the noise of the FF microphone 137. The second part of the calibration process can estimate the headphone transfer function by playing a specific signal to the headphones and recording the signal of the FB microphone 137. After calculating the optimal FF ANC filter, the DSP 110 can program the coefficients of the RAP 120.

  FIG. 2 is a schematic diagram illustrating an exemplary RAP I / O 200 that can be applied to a RAP, for example, RAP 120. RAP I / O 200 includes a processor peripheral bus 241. The processor peripheral bus 241 provides control and configuration parameters (e.g., user input, commands, calculated noise filters, compression filters, environment awareness filters, and / or other filters described herein) from the DSP. , Control and configuration parameters 141). RAP I / O 200 also includes an input for receiving an audio signal (eg, music) 243 from the DSP. The audio signal 243 may be substantially similar to the audio signal 143. RAP I / O 200 further includes an input for receiving noise signal 244. The noise signal 244 may be substantially similar to the noise signal 144. Noise signal 244 is shown as four inputs. The reason is that when the FF and FB microphones are used for each of the left and right headphones, there are four noise signals 244. However, any number of noise signals 244 may be used. RAP I / O 200 includes outputs for outputting output signal 245, anti-noise signal 246, and intermediate signal 242. Anti-noise signal 246 may be generated based on a noise filter received via processor peripheral bus 241 and a noise signal 244 received from a corresponding microphone. The anti-noise signal 246 is forwarded to an amplifier controller that supports control of the DAC amplifier, which can reduce clipping and associated noise disturbances. Output signal 245, which is substantially similar to output signal 145, may include anti-noise signal 246 mixed with equalized audio based on audio signal 243. The output signal 245 may be transferred to the left and right speakers and played back to the user. The intermediate signal 242 may be performed by a partially equalized audio signal, an anti-noise signal 246, a partially generated anti-noise signal, a RAP state, a compressed state, a currently used filter, and / or a RAP. Other RAP information indicating the audio processing performed may be included. By forwarding the intermediate signal 242 as feedback to the DSP, the DSP can take into account current RAP operating parameters when making changes to the RAP function. Thus, the intermediate signal 242 allows the DSP to dynamically modify the RAP configuration to improve performance and advanced control. To match the intermediate signal 242 with the processing frequency used by the DSP, a portion of the intermediate signal 242 may be resampled by passing it through a decimation filter. Other intermediate signals 242 (eg, slowly changing signals such as signal levels and processor gains) are transferred to the DSP via a register interface for periodic sampling. Note that RAP I / O 200 may include other inputs and / or outputs. Although the main functional I / O was described using RAP I / O 200, it is not intended to be exhaustive.

  FIG. 3 is a schematic diagram illustrating an exemplary sound processing network 300 for sharing compressor state. Network 300 includes DSP 310 and RAP 320. DSP 310 and RAP 320 may be substantially similar to DSP 110 and RAP 120, respectively. Other components have been omitted for clarity. RAP 320 includes an adjustable amplifier 326. Adjustable amplifier 326 may be any circuit that can change the signal gain to a target value set by RAP 320. As described above, the RAP 320 generates the anti-noise signal 342 based on the filter from the DSP 310 and the noise signal from the microphone. Adjustable amplifier 326 cancels noise by amplifying anti-noise signal 342 to a sufficient value (eg, after conversion by a DAC and associated amplifier). The RAP 320 further includes a RAP compressor circuit 325. RAP compressor circuit 325 may be any circuit configured to control tunable amplifier 326. Specifically, the RAP compressor circuit 325 controls the adjustable amplifier 326 to reduce disturbance of the anti-noise signal 342 due to clipping or the like. The RAP 320 further includes a compression status register 323. Compression status register 323 may be any read / write memory component. The compression state register 323 stores the compression state, and the RAP compressor circuit 325 controls the adjustable amplifier 326 based on the compression state.

  The RAP compressor circuit 325 and the adjustable amplifier 326 can be used to mitigate sudden changes in the value of the anti-noise signal 342. For example, the RAP compressor circuit 325 and the adjustable amplifier 326 can mitigate the spike in the value of the anti-noise signal 342 (and associated signal disturbances) due to the slam closing of the car door, but can be noisy from quiet rooms. Allows the anti-noise signal 342 to increase continuously due to movement into the room. To determine the adjustment of adjustable amplifier 326, RAP compressor circuit 325 considers the compression state stored in compression state register 323. The compression state may include a peak signal estimate of the anti-noise signal 342, instantaneous gain, target gain, attack parameters, release parameters, peak attenuation parameters, maintenance parameters, and / or root mean square (RMS) parameters. The peak signal estimate includes an estimate of the maximum expected value of the anti-noise signal 342. Utilizing the peak signal estimate to determine an appropriate amount of amplification can prevent any portion of the anti-noise signal 342 from being amplified outside the DAC amplifier (eg, causing clipping). it can. The instantaneous gain indicates the current gain provided by the tunable amplifier 326 at a particular moment, and the target gain indicates the gain that the tunable amplifier 326 must achieve to adjust for the signal change. The attack parameter indicates the speed at which the gain is increased without causing signal disruption. The release parameter indicates the speed at which the gain is reduced without causing signal disturbance. The maintenance parameter indicates the time for which the increased gain is maintained after the anti-noise signal 342 returns to a normal value, for example, in preparation for the possibility of another large noise. The peak decay parameter indicates the amount by which the anti-noise signal 342 must change from its peak value before the anti-noise signal 342 is considered to have returned to a normal value due to the maintenance parameter. Additionally or alternatively, tunable amplifier 326 may be tuned based on the RMS of anti-noise signal 342 to reduce clipping.

  RAP 320 operates much faster than DSP 310, but is limited to very simple compression algorithms. Therefore, the DSP 310 includes the DSP compressor 311. The DSP compressor 311 is a programmable circuit that can take into account the compression state of the RAP 320 and apply a complex compression algorithm to the compression state to determine a more accurate setting of the amplifier 326 that can be adjusted on a slower time scale. . Thus, the DSP 310 is configured to receive the current compression state stored in the compression state register 323 from the RAP 320. This data may be communicated via the output of the intermediate signal (eg, the intermediate signal 242) and / or the path of the RAP state signal (eg, the RAP state 142). The DSP compressor 311 can determine a new compression state based on the noise signal and the current compression state. Next, the DSP compressor 311 can transfer the new compression state to the RAP to support control of the adjustable amplifier 326. For example, the DSP compressor 311 can directly program the RAP 320 to perform compression by transferring the new compression state to the compression state register 323.

  FIG. 4 is a schematic diagram illustrating an exemplary sound processing network 400 for performing equalization of audio input. The sound processing network 400 includes a DSP 410 and a RAP 420, which may be substantially similar to each of the DSPs 110 and 310 and RAPs 120 and 320. As described above, DSP 410 may generate audio signal 443 for use by RAP 420 based on audio input 448. The DSP 410 can generate the audio signal 443 using the first equalizer 412. An equalizer is any circuit for adjusting the frequency response of a network for practical or aesthetic reasons. For example, the first equalizer 412 can customize the audio signal 443 by adjusting bass, treble, etc. of the audio according to the frequency response of the network 400.

  It is difficult to cancel the noise by applying an anti-noise signal at the same time as playing the sound to the user. Specifically, an FB microphone in the user's ear canal may record all or part of the audio signal 443 as noise. In this case, the RAP 420 generates an anti-noise signal and cancels a part of the audio signal 443. For example, the anti-noise signal cancels out some low-frequency audio of the audio signal 443, resulting in incorrect performance of the headphones. To address this problem, the DSP 410 includes a second equalizer 413. Second equalizer 413 is substantially similar to first equalizer 412, but is used for a different purpose. DSP 410 and / or second equalizer 413 model the frequency response of network 400. The second equalizer 413 then uses the model to generate an expected output signal 449 based on the audio input 448 and the frequency response of the sound processing network 400. Expected output signal 449 is, in effect, a copy of audio signal 443 modified in accordance with the expected effect of the circuitry of network 400. If there is no audio, the ANC process will try to zero out the noise. By transferring the expected output signal 449 to the RAP 420, the ANC process can set the expected output signal 449 as a reference point. Thus, the ANC process can make the output signal from RAP 420 an expected output signal 449 instead of zero. This method may reduce / eliminate the ANC effect of the audio signal 443.

  Therefore, the RAP 420 receives the audio signal 443 from the DSP 410. The RAP 420 then mixes the audio signal 443 with the anti-noise signal. Further, when generating the anti-noise signal, the RAP 420 reduces the cancellation of the audio signal due to the anti-noise signal by setting the expected output signal 449 as a reference point.

  FIG. 5 is a schematic diagram illustrating an exemplary RAP architecture 500. RAP architecture 500 may be used for RAPs 120, 320 and / or 420, for example. The RAP architecture 500 employs a biquad engine 524, a multiply accumulator 525, a data register 522, and a biquad memory 521. These components generate an output signal, eg, output signal 145, by filtering the input using biquadratic coefficients 527, gain coefficients 526, and feathering / compression gain coefficients 523.

  The biquad engine 524 is a circuit for generating a digital filter having two poles and two zeros. The poles are the roots of the denominator of the system transfer function polynomial, and the zeros are the numerator of the transfer function polynomial. In other words, the poles push the signal being filtered towards infinity, and the zeros push the signal being filtered towards zero. Note that if the poles are non-zero, such filters have an infinite impulse response (IIR). Such a filter is called a biquad filter. Its meaning refers to the concept that the transfer function of a filter is the ratio of two quadratic functions. Bi-secondary engine 524 operates at a higher frequency than the signals processed by bi-secondary engine 524. Thus, biquad engine 524 may be applied multiple times to a single signal sample and / or may be applied to different portions of the signal in different ways. Because the bi-secondary engine 524 is programmable, it can create various topologies of the processes described below. Although the RAP architecture 500 is described as using a biquad filter, other filter architectures, i.e., filters with poles and zeros other than two poles and two zeros, depending on the particular implementation details. Can be used to replace the biquad filter.

  The multiplication accumulator 525 is a circuit for adding and / or multiplying a value. For example, a multiply accumulator 525 may be used to scale the signal and / or the signal portion. In addition, the multiply accumulator 525 can be used to calculate a weighted sum of multiple signals and / or signal portions. Multiply accumulator 525 may accept output from biquad engine 524 and vice versa. Data register 522 may be any memory element for storing data. Specifically, data register 522 can store signals such as the output from biquad engine 524 and / or multiply accumulator 525. Accordingly, biquad engine 524, multiply accumulator 525, and data register 522 cooperate to perform a mathematical and / or other special digital signal modification process to sample audio signal 543 and / or noise signal 544. Can be applied iteratively. The audio signal 543 and the noise signal 544 may be substantially similar to the audio signal 143 and the noise signal 144, respectively.

  The bi-secondary state memory 521 is a memory module for storing the current bi-secondary state, for example, a register. The biquad engine 524 is programmable to operate as a finite state machine. Bi-secondary state memory 521 stores data indicating available and / or current states of bi-secondary engine 524. Bi-secondary engine 524 can read data from bi-secondary state memory 521 and store data in bi-secondary state memory 521.

  In summary, state data from the bi-quadratic state memory 521 can be used to program the bi-quad engine 524 and the multiplier accumulator 525 to implement various topologies. The data of the intermediate signal may be stored in the data register 522. RAP architecture 500 can receive control and configuration parameters 541, which can be substantially similar to control and configuration parameters 141. Control and setting parameters 141 include a noise filter encoded based on biquadratic coefficients 527 and gain coefficients 526. Biquad engine 524 modifies the shape of the signal being processed (eg, audio signal 543 and / or noise signal 544) based on biquad coefficients 527 that may be stored in local memory as received from the DSP. I do. Further, multiply accumulator 525 increases the gain of the signal being processed (eg, audio signal 543 and / or noise signal 544) based on a gain factor 526 that may be stored in local memory as received from the DSP. /change.

  In some cases, the gain coefficients need to be feathered. Feathering means a gradual change from a first value to a second value. Multiply accumulator 525 can function as a feathering unit by porting the feathering coefficients received from the input of feathering / compression gain 523. For example, the multiply accumulator 525 can implement three feathering units on the left channel and three feathering units on the right channel. In another example, multiply accumulator 525 may implement six feathering units for each channel.

  Multiplier accumulator 525 can receive the compression state from the input of feathering / compression gain 523. This compressed state may be substantially similar to compressed state 323, stored in local memory, or received from a DSP. Multiply accumulator 525 can function as a compressor (eg, a non-linear processor) that can change the gain applied to the signal if the signal is too strong. By using the multiply accumulator 525 to dynamically lower the gain of the signal stream, clipping can be avoided. For example, if the anti-noise is too strong for the DAC, applying a compressor to the anti-noise signal can temporarily reduce the gain. This temporarily reduces the ANC strength, but prevents the occurrence of unpleasant disturbances due to signal clipping. Multiply accumulator 525 can implement three feathering units on the left channel and three feathering units on the right channel. In another example, multiply accumulator 525 may implement six feathering units for each channel.

  The RAP architecture 500 can implement one or more programmable biquad filters by using different coefficients across multiple states of the finite state machine. These biquad filters implement a noise filter from the DSP to generate an anti-noise signal. Also, RAP architecture 500 can mix anti-noise / noise signal 544 with audio signal 543. Further, the RAP architecture 500 can apply filters to the audio signal 543 as needed.

  FIG. 6 is a schematic diagram illustrating another exemplary RAP architecture 600. RAP architecture 600 is a particular implementation of RAP architecture 500. For clarity, RAP architecture 600 has been shown to operate to generate ANC, omitting audio signal processing. RAP architecture 600 includes a multiplier accumulator 625, which is a circuit for multiplying and / or adding signal data. RAP architecture 600 also includes an accumulator register 622, which is a memory circuit for storing the output of multiply accumulator 625. Multiply accumulator 625 and accumulator register 622 may implement multiply accumulator 525. Further, RAP architecture 600 includes bi-quad engine 624 and bi-quad output register 628. Bi-secondary engine 624 and bi-secondary output register 628 may implement bi-secondary engine 524. The biquad engine 624 is a circuit for implementing a filter, and the biquad output register 628 is a memory for storing the result of the calculation by the biquad engine 624. RAP architecture 600 also includes a bi-secondary memory 621, which may be a memory unit for storing partial results from bi-secondary engine 624. Bi-secondary memory 621 can implement bi-secondary state memory 521.

  As shown, these components are linked together by multiplexers (MUX) 661, MUX 662 and MUX 663, and are linked to external local memory and / or remote signals (eg, from a DSP). As shown, these components include a feathering coefficient 623, a multiplication coefficient 626, and a biquadratic coefficient 627, which may be substantially similar to the feathering / compression gain 523, gain coefficient 526, and biquadratic coefficient 527, respectively. Can be received. These components can receive a noise signal 644 from a microphone / speaker to perform ANC. Noise signal 644 may be substantially similar to noise signal 144. These components can also receive a cycle index 647. The cycle index 647 is data indicating the current position in the RAP duty cycle. As shown, the various signals, indices and coefficients are routed to corresponding components via MUXs 661-663.

  In operation, the cycle index 647 is used to select a biquadratic coefficient 627 for the corresponding state. Biquad coefficient 627 and / or cycle index 647 are forwarded to biquad engine 624 and applied to noise signal 644. The state information can be obtained from the bi-secondary memory 621. Also, the partial results may be stored in biquad memory 621 and / or fed back to biquad coefficients 627 and applied to the next state. The final result may be stored in biquad output register 662 for output to multiply accumulator 625. Further, the output from the bi-secondary output register 662 can be fed back to the bi-secondary engine 624. Also, the output from accumulator register 622 can be forwarded back to biquad engine 624. Further, noise signal 644 may bypass biquad engine 624 and pass directly to multiply accumulator 625.

  Further, the multiplication coefficient 626 in the corresponding state is selected using the cycle index 647. Multiplier coefficients 626, feather coefficients 623 and / or cycle index 626 are transferred to multiply accumulator 625 and applied to various inputs. Multiplier accumulator 625 may receive as inputs the output of biquadratic output register 662, noise signal 644, and / or the output of multiply accumulator 625. In other words, the output of multiply accumulator 625 can be fed back to the input of multiply accumulator. Once the coefficients are applied to the inputs based on the corresponding states, the output of multiply accumulator 625 is stored in accumulator register 622 for output to other components. The output of accumulator register 622 and / or the output of biquadratic output register 628 may be forwarded to the speaker as the output of RAP architecture 600. As described below, the interconnections of the RAP architecture 600 can implement different topologies applied to different audio processing schemes by programming the components.

  FIG. 7 is a schematic diagram illustrating an exemplary programmable topology 700 of RAPs, such as RAPs 120, 320 and / or 420, implemented in accordance with RAP architectures 500 and / or 600. Topology 700 is configured to output audio signals and provide ANC. Topology 700 receives a first audio signal 743 (audio 1) and a second audio signal 753 (audio 2). The audio signals 743 and 753 may be substantially similar to the audio signal 143 and may include audio provided to the left and right ears, respectively. In some examples, audio signals 743 and 753 may be expected output signal 449 and audio signal 443, respectively. Topology 700 also receives FB microphone signal 744 and FF microphone signal 754, which may be substantially similar to noise signal 144. An audio signal for ANC is generated as an output 754 using noise signals including the audio signals 743 and 753 and the FB microphone signal 744 and the FF microphone signal 754.

  The topology uses an amplifier 729 to amplify the first audio signal 743, the second audio signal 753, and the FB microphone signal 744. Such an amplifier may be implemented by a multiply accumulator, such as multiply accumulator 525, using gain factors in the first three states. Second audio signal 753 and FB microphone signal 744 are mixed by mixer 725. Mixer 725 may be implemented by a multiply accumulator in a fourth state. The output of the mixer is transferred through a series of biquad filters 724, in this example eight directly connected biquad filters 724. Biquad filter 724 may be implemented by multiply accumulator and biquad engine 524 using a corresponding set of biquad coefficients 527 (eg, across eight states). At the same time, the FF microphone signal 754 is transferred through a series of biquad filters 724, in this example eight biquad filters 724. The second audio signal 753 and the FB microphone signal 744 mixed with the FF microphone signal 754 are amplified by a respective amplifier 729 and mixed by a mixer 725 (eg, each into a corresponding state of a multiplying accumulator). Implemented). Next, the mixed FF microphone signal 754, second audio signal 753, and FB microphone signal 744 are transferred to the feathering amplifier 726 and feathered. This may be implemented by a multiply accumulator, for example, using the feathering coefficients from the feathering / compression gain 523. The feathering results are mixed by mixer 725 (which may be implemented, for example, by a multiplier-accumulator) to produce output 745.

  As can be seen from the above discussion, the components of the biquad engine and the accumulator can apply different operations on samples from each signal in different states. The biquad engine and the multiply-accumulator implement the topology 700 across the various states and generate an output 745 by performing corresponding operations on the samples. After generating the output 745 for one set of samples, another output 745 is generated by taking another set of samples and modifying them in various states. Further, the topology 700 can be changed by reprogramming the states of the biquad engine and multiply accumulator according to the relevant coefficients.

  FIG. 8 is a schematic diagram illustrating another example programmable topology 800 of RAPs, such as RAPs 120, 320 and / or 420, implemented in accordance with RAP architectures 500 and / or 600. The topology 800 is created by reprogramming the topology 700. Topology 800 is configured to provide adaptive ANC, environment recognition and sidetone enhancement. Thus, upon receiving input from a user, topology 800 can be obtained by reconfiguring topology 700 to include environment awareness and sidetones. Environment recognition operates to emphasize a particular frequency band. For example, by emphasizing the frequency bands associated with human speech, the ANC can cancel noise and emphasize speech as part of the conversation. Sidetone refers to the voice of the user. Thus, by providing sidetone enhancement using the topology 800, the user can clearly hear his or her own voice. Thus, while the topology 800 can reduce environmental noise, it allows the user to clearly hear not only his own voice but also others' voices. Thus, using the topology 800, a pair of headphones can be converted to a hearing enhancement device.

  Topology 800, like topology 700, uses a biquad filter 824 that can be implemented by a biquad engine, such as biquad engine 524. The topology 800 also uses an amplifier 829, a mixer 825, and a feathering amplifier 826, which, like the topology 700, can be implemented by a multiply accumulator, such as the multiply accumulator 525. The topology 800 includes a first audio signal 843 (Audio 1), a second audio signal 743, a second audio signal 753, a FB microphone signal 744, and a FF microphone signal 754, each of which is substantially similar to the FF microphone signal 754. The audio signal 853 (audio 2), the FB microphone signal 844, and the FF microphone signal 854 are received.

  The FF microphone signal 854 is used for performing environment recognition. For example, the biquadratic filter 824 on the path of the FF microphone signal 854 functions as an environment recognition filter. Therefore, when the topology 800 is generating an anti-noise signal, a predetermined frequency band of the noise signal can be enhanced by applying an environment recognition filter to the path of the FF microphone signal 854. Thereby, a predetermined frequency band, for example, an utterance band can be strengthened. The path of the FF microphone signal 854 can transfer an anti-noise signal having a predetermined frequency band enhanced to a speaker and output it to the user as an output 845.

  Further, the topology 800 uses a first audio microphone signal 848 (audio microphone 1) and a second audio microphone signal 858 (audio microphone 2). These signals may be recorded by a microphone arranged to record the user's voice, for example, microphone 137. For example, such a microphone may be mounted on a lapel clip, for example, attached to headphones and placed on the user's chest. Thus, the first audio microphone signal 848 and the second audio microphone signal 858 can include samples of sidetones (eg, a user's voice).

  Functionally, each of the FB microphone signal 844 and the first audio microphone signal 848 is transferred via a biquad filter 824 and an amplifier 829. Further, the second audio microphone signal 858 and the second audio signal 853 are transferred via the amplifier 829. As shown, these signals are mixed by mixer 825. The result is transferred through a set of biquad filters 824, in this case five series filters, and another amplifier 829. These signals include sidetones, the FB portion of the ANC, and the second portion of the audio signal.

  On the other hand, the FF microphone signal 854 including the FF portion and the environment recognition portion of the ANC is transferred via the feathering amplifier 826. Using this feathering amplifier 826, the environment recognition and ANC mode can be changed lightly. Next, the FF microphone signal 854 is transmitted in parallel via a biquadratic filter 824, in this case three serial filters and five serial filters. The result is amplified by amplifier 829 and mixed by mixer 825. A part of the mixing result is transferred via the biquad filter 824, the amplifier 829, and the second feathering amplifier 826. Another part of the mixing result is transferred in parallel, bypassing these components. The paths are mixed together by mixer 825. The second feathering amplifier 826 uses a compressor to enable strong FF ANC without signal clipping.

  The result of the path of the FF microphone signal 854 is then amplified by the amplifier 829 before being mixed into the signal path including the sidetone, the FB portion of the ANC, and the second portion of the audio signal. As shown, the FF microphone signal 854 path is mixed by mixer 825 before and after five biquad filters 824. The result of such a signal passes through another feathering amplifier 826 used to turn ANC on and off. The feathering amplifier 826 can further reduce clipping by applying a digital compressor. Further, the first audio signal is amplified by amplifier 829 and mixed with the remainder of the signal by mixer 825. This mixes the audio signal, the FF anti-noise signal, the FB anti-noise signal, sidetones, and environmental recognition enhancements all together to produce an output 845 that is played back to the user via speakers.

  FIG. 9 is used by a biquad engine, such as biquad engine 524 and / or 624, for example, and may be applied to the noise, anti-noise, audio, and / or other signals disclosed herein. FIG. 3 is a schematic diagram illustrating a structure of a biquad filter 900. In general, biquad filters are mathematically described according to Equation 1 below.

Where x [n] is the input to the biquad filter, y [n] is the output from the biquad filter, and b ₀ , b ₁ , b ₂ , a ₁ and a ₂ are , Bilinear coefficients, for example, biquadratic coefficients 527 and / or 627. Therefore, the function of the biquad filter 900 can be changed by changing these coefficients.

Instead, biquad filter 900 uses different coefficients. Specifically, as shown, the biquadratic filter 900 has gain coefficients b ₀ (973), −c ₁ (975), −c ₂ (976), d ₁ (974), and d ₂ (978). ). Gain factor 973 may be implemented by an adjustable amplifier. Other gain factors are mathematically defined by Equations 2-5 below, with reference to Equation 1.

The biquad filter 900 also uses a multiplier 972, which may be implemented by a multiply accumulator. In operation, biquad filter 900 receives an input. This input is forwarded to the output via mixer 982 and gain coefficient b ₀ (973). The input is transferred to the past state block 971 via another mixer 981 and stored in the memory. In the next cycle / state, the output of past state block 971 is transferred to mixer 983 via gain coefficient d ₁ (974), to mixer 984 via gain coefficient −c ₁ (975), and to mixer 985 To another past state block 972. In another state, the output of past state 972 is transferred to mixer 983 via gain factor d ₂ (978). The mixer 983 mixes the output of the past state 972, the gain coefficient d ₂ (978), the output of the past state 971, and the gain coefficient d ₁ (974). The result is transferred to mixer 982 and mixed with the input. The output of the past state 972 is transferred to the mixer 984 via the gain coefficient −c ₂ (976). Therefore, the output of the past state 972 and the gain coefficient −c ₂ (976) are mixed with the output of the past state 971 and the gain coefficient −c ₁ (975). The result is transferred to mixer 981. Mixer 981 mixes the result from mixer 984 with the input and feeds back to past state 971. The biquadratic filter 900 uses a switch 977 that applies a gain of “0” or a gain of “1”. When the switch 977 sets the gain of “1”, the output of the past state 972 can be fed back to the past state 972 via the mixer 985. When the switch 977 sets “0”, the biquadratic filter 900 is converted to a so-called direct biquadratic filter by changing all the coefficients according to Equation 1.

  As shown, the input changed in the first state is mixed with the input changed in the second state, and then mixed with the input in the third state. Thus, the input signal samples continually modify further input samples that are received later.

  It should be noted that the error source of the biquad filter is quantization. Quantization occurs, for example, when storing signal samples in past states 971 and / or 972. Specifically, if the memory used to store the samples is not sufficient to store the samples at full resolution, quantization will occur as a result of rounding errors. As mentioned above, biquad filters use poles and zeros. Direct biquad filters attenuate the signal by applying zeros, preserve the signal responsible for quantization, and then amplify the signal by applying poles. This technique amplifies errors associated with quantization. Such a direct biquad filter typically uses more bits than the biquad filter 900 to obtain a reasonable signal-to-noise ratio (SNR). In contrast, biquad filter 900 amplifies the signal, preserves and quantizes the signal, and then attenuates the signal. With this approach, quantization errors are attenuated rather than amplified. As a result, the biquad filter 900 can achieve a 60 dB lower SNR than a direct biquad using a similar number of bits in the past state memory. Alternatively, the biquad filter 900 can operate with about 10 bits less memory to achieve a similar SNR, saving substantial space.

The order of operation of the biquad filter 900 can be determined by the coefficients. Specifically, b ₀ (973), d ₁ (974), and d ₂ (978) apply a zero, and −c ₁ (975) and −c ₂ (976) apply a pole. As shown in FIG. 9, before being quantized by past states 971 and 972, the signal passes through an amplifier that always applies poles (−c ₁ (975), −c ₂ (976)). The output of this state is fed back to the system or used by amplifiers that apply zeros (eg, zeros b ₀ (973), d ₁ (974), and d ₂ (978)) for use in later states. Output via

  In other words, biquadratic filter 900 uses poles to amplify a portion of the noise / anti-noise signal sample. Biquadratic filter 900 also uses zeros to attenuate some of the noise / anti-noise signal samples. Further, biquadratic filter 900 uses a filter register to store the quantization of the noise / anti-noise signal samples. Further, biquad filter 900 is configured to amplify the sample before quantizing the sample and then attenuate the sample.

  The goal of the biquadratic design is to minimize the requirements by reducing the storage size and current while achieving the desired performance given the input signal type and the target filter. As mentioned above, the target frequency of the biquadratic filter used herein is typically in the audio band (eg, less than 20 kHz), which is significantly less than the sampling rate (eg, less than 1 MHz). In this scenario (eg, where the center frequency is much lower than the sampling rate), the performance of the biquad filter 900 significantly exceeds the performance of the biquad design. As an example, when operating at about 6.144 MHz to implement a peak filter with a quality factor (Q) of 1 at 250 Hz and a gain of 40 dB, the biquad filter 900 may be a direct biquad with the same number of bits. Approximately 60 dB lower noise can be generated. As a result, about 10 bits can be saved.

Another feature is that the biquad filter 900 does not need to apply a multiplier directly to the input signal. This results in a design that can be easily pipelined. Further, the multiplication by b ₀ (973) is located near the output. Therefore, biquadratic filter 900 functions as a filter following the final gain stage. This is useful when multiple biquads are used in series. In that case, the b ₀ (973) multiplication can be combined into a signal multiplication step. Thus, for N biquads, direct biquad requires 5N multiplications. In contrast, biquad filter 900 uses only 4N + 1 multiplications. Providing a multiplier at the output of the serial cascade may be particularly useful in the case of a RAP hardware architecture.

  FIG. 10 includes a RAP architecture 500 and / or 600 that uses a topology such as 700 and / or 800 that includes an I / O such as RAP I / O 200 and a biquad filter such as biquad filter 900. 4 is a flowchart illustrating an example method 1000 of operating a sound processing network, such as networks 100, 300 and / or 400, including a RAP. In other words, method 1000 may be implemented using various combinations of the components shown in the various figures described above.

  At block 1001, the DSP generates an audio signal based on the audio input. Further, the DSP generates an expected output signal based on the speech input and the frequency response of the sound processing network. Next, as shown in network 400, the audio signal and the expected output signal are transmitted from the DSP to the RAP.

  At block 1003, the DSP receives the noise signal. The noise signal is received from at least one microphone. The DSP generates a noise filter based on the noise signal. Also, as shown in network 100, the DSP sends a noise filter from the DSP to the RAP. As described above, the DSP operates at the first frequency, and the RAP operates at a second frequency higher than the first frequency.

  At block 1005, the RAP controls an adjustable amplifier using the current compression state in the RAP to adjust the anti-noise signal. As shown in network 300, the current compression state used by the RAP is transmitted from the RAP to the DSP. The DSP determines a new compression state based on the noise signal and the current compression state. The DSP sends a new compression state to the RAP to support adjustable amplifier control. The new compression state may include a peak signal estimate, instantaneous gain, target gain, attack parameters, release parameters, peak attenuation parameters, maintenance parameters, RMS of the anti-noise signal, or a combination thereof.

  At block 1007, the RAP receives an audio signal, an expected output signal, a noise filter, and / or a new compression state from the DSP and a noise signal from a microphone (eg, FF and / or FB microphone). I do.

  At block 1009, the RAP generates an anti-noise signal used for ANC based on the noise signal and the noise filter. Furthermore, when generating an anti-noise signal, the RAP sets the expected output signal as a reference point, thereby reducing the cancellation of the audio signal due to the anti-noise signal. An anti-noise signal can be generated at the RAP by configuring a biquad filter that is programmable to implement a noise filter from the DSP. For example, as shown in the biquadratic 900, a biquadratic filter can amplify the samples of the anti-noise signal, quantize the samples of the anti-noise signal, and then attenuate the samples of the anti-noise filter.

  At block 1011, when generating an anti-noise signal as described with respect to topology 800, an environment-aware filter is applied to the RAP to enhance a predetermined frequency band of the noise signal. Thereby, a predetermined frequency band, for example, a frequency band related to voice can be enhanced. In some examples, additional filters may be applied to add sidetones.

  At block 1013, the RAP mixes the audio signal with the anti-noise signal. The RAP transfers the obtained signal to a speaker and outputs it to the user. In some cases, the resulting signal may include speech, anti-noise, sidetones, an environment-aware signal with an enhanced predetermined frequency band, and / or other features described herein.

  At block 1015, the RAP transfers the anti-noise signal to a DAC amplifier controller to support adjustment of the DAC amplifier based on the anti-noise signal level, thereby reducing clipping and other disturbances. It should be noted that the method 1000 described above seeks to account for the simultaneous operation of all features disclosed herein. Thus, the method 1000 includes many selection steps because not all functions need to be active at all times. Further, method 1000 may not always operate in the order shown, as it may operate constantly.

  Examples of the present disclosure can operate on specially-manufactured hardware, firmware, digital signal processors, or specially programmed general-purpose computers that include a processor that operates in accordance with programmed instructions. The term "controller" or "processor" as used herein is intended to include microprocessors, microcomputers, application specific integrated circuits (ASICs), and special purpose hardware controllers. One or more aspects of the present disclosure relate to computer-usable data and computer-executable instructions, such as one or more processor modules (including monitoring modules) or one or more program modules executed by other devices (eg, a computer). Program product). Generally, program modules include routines, programs, objects, components, data structures, etc. that perform particular tasks or implement particular abstract data types when executed by a processor in a computer or other device. The computer-executable instructions are stored on a non-transitory computer-readable medium, such as a random access memory (RAM), a read only memory (ROM), a cache, an electrically erasable programmable read only memory (EEPROM), a flash memory, or other Memory, compact disk read only memory (CD-ROM), digital video disk (DVD), or other optical disk storage device, magnetic cassette, magnetic tape, magnetic disk storage device, or other magnetic storage device, implemented in any technology Other volatile or non-volatile media, removable or non-removable media. Computer-readable media excludes the signals themselves and any forms of temporary signal transmission. Also, the functions can be fully or partially implemented with firmware or hardware equivalents, such as, for example, integrated circuits, field programmable gate arrays (FPGAs), and the like. Certain data structures may be used to more effectively implement one or more aspects of the present disclosure. Such data structures are considered to be within the scope of the computer-executable instructions and computer-usable data described herein.

  Aspects of the present disclosure operate with various modifications and alternatives. Particular aspects are illustrated by way of example in the drawings and are described in detail below. However, the examples disclosed herein are provided for clarity of explanation, and unless otherwise indicated, the scope of the general concepts disclosed is not limited to the specific examples described herein. It is not intended to be limited to Accordingly, the present disclosure is intended to cover all modifications, equivalents, and alternatives of the aspects set forth in light of the accompanying drawings and claims.

  Reference herein to embodiments, implementations, and examples indicate that the items described may include particular features, structures or characteristics. However, not all disclosed aspects need necessarily include the particular feature, structure, or characteristic. Moreover, such expressions are not necessarily referring to the same embodiment unless otherwise specified. Furthermore, when a particular feature, structure, or characteristic is described in relation to a particular aspect, such feature, structure or characteristic is not described, whether explicitly described in relation to the other aspect. , Can be applied to another embodiment.

Examples The following provides illustrative examples of the technology disclosed herein. Embodiments of the present technology may include any one or more or any combination of the examples described below.

  Example 1 includes a sound processing network. The sound processing network includes a digital signal processor (DSP) operating at a first frequency, the DSP receiving a noise signal from at least one microphone and generating a noise filter based on the noise signal. The sound processing network comprises a real-time sound processor (RAP) operating at a second frequency higher than the first frequency, the RAP receiving a noise signal from a microphone, receiving a noise filter from a DSP, and a noise signal. And generating an anti-noise signal for use in active noise cancellation (ANC) based on the noise filter.

  Example 2 includes the sound processing network of Example 1, wherein the RAP reduces the disturbance of the anti-noise signal by controlling the adjustable amplifier for amplifying the anti-noise signal and the adjustable amplifier. And a compressor circuit.

  Example 3 includes the sound processing network of Example 2, wherein the RAP further includes a compression state register for storing a compression state, and the compressor circuit further controls the adjustable amplifier based on the compression state. .

  Example 4 includes the sound processing network of Example 3, wherein the compression state includes a peak signal estimate, an instantaneous gain, a target gain, an attack parameter, a release parameter, an attenuation parameter, a maintenance parameter, or a combination thereof.

  Embodiment 5 includes the sound processing network of Embodiment 3, and the compression state includes the root mean square (RMS) of the anti-noise signal.

  Example 6 includes the sound processing network of Examples 1-4, wherein the DSP receives the current compression state from the RAP, determines a new compression state based on the noise signal and the current compression state, and Supports adjustable amplifier control by transferring the new compression state to the RAP.

  Example 7 includes the sound processing network of Examples 1-6, wherein the RAP implements a noise filter from a DSP and includes one or more programmable biquad filters to generate an anti-noise signal. .

  Example 8 includes the sound processing network of Example 7, wherein the biquad filter amplifies a portion of the sample of the anti-noise signal using one or more poles and uses one or more zeros. Attenuate some of the samples of the anti-noise signal and store the quantization of the samples of the anti-noise signal using a filter register, the biquad filter amplifies the samples before quantizing the samples, Attenuate the sample.

  Example 9 includes the sound processing network of Examples 1 to 8, wherein the microphone is a feedforward microphone, and the RAP further applies an environment recognition filter to generate the anti-noise signal, By enhancing the predetermined frequency band, an enhanced predetermined frequency band is generated, and an anti-noise signal including the enhanced predetermined frequency band is transferred to a speaker and output to a user.

  Example 10 includes the sound processing network of Examples 1-9, wherein the delay between receiving the noise signal sample from the microphone and transferring the corresponding anti-noise signal sample to the speaker is less than 100 microseconds.

  Example 11 includes the sound processing network of examples 1 to 10, wherein the DSP further generates a sound signal based on the sound input and an expected output based on the sound input and the frequency response of the sound processing network. Generating a signal, the RAP further receiving an audio signal from the DSP, mixing the audio signal with the anti-noise signal, and setting the expected output signal as a reference point when generating the anti-noise signal. Thereby, cancellation of the audio signal due to the anti-noise signal is reduced.

  Example 12 includes the sound processing network of Examples 1-11, wherein the RAP transfers the anti-noise signal to a digital-to-analog converter (DAC) amplifier controller to adjust the DAC amplifier based on the anti-noise signal level. Further configured to support.

  Example 13 includes a method. The method includes receiving a noise signal at a digital signal processor (DSP) operating at a first frequency, wherein the noise signal is received from at least one microphone and generates a noise filter at the DSP based on the noise signal. Transmitting a noise filter from the DSP to a real-time sound processor (RAP) operating at a second frequency higher than the first frequency; receiving a noise signal from the microphone at the RAP; Generating an anti-noise signal for use in active noise cancellation (ANC) at the RAP based on the noise filter.

  Example 14 includes the method of Example 13, wherein the method adjusts the anti-noise signal by controlling an adjustable amplifier using the current compression state at the RAP, and changes the current compression state to the RAP. From the DSP to the DSP, determining a new compression state based on the noise signal and the current compression state, and transmitting the new compression state from the DSP to the RAP, thereby providing an adjustable amplifier. Supporting the control.

  Example 15 includes the method of Example 14, wherein the compression state includes a peak signal estimate of the anti-noise signal, an instantaneous gain, a target gain, a root mean square (RMS), or a combination thereof.

  Example 16 includes the method of Examples 13-15, wherein the anti-noise signal is generated at the RAP by configuring one or more programmable biquad filters to implement a noise filter from the DSP. Is done.

  Example 17 includes the method of example 16, wherein the biquad filter amplifies the samples of the anti-noise signal, then quantizes the samples of the anti-noise signal, and then attenuates the samples of the anti-noise filter.

  Example 18 includes the method of Examples 13-17, wherein the method enhances a predetermined frequency band of the noise signal by applying an environment-aware filter at the RAP when generating the anti-noise signal. Generating the enhanced predetermined frequency band, and transmitting an anti-noise signal including the enhanced predetermined frequency band to a speaker and outputting the signal to a user.

  Example 19 includes the method of Examples 13-18, wherein the method generates an audio signal at the DSP based on the audio input and expects at the DSP based on the frequency response of the audio input and the audio processing network. Generating an audio signal, transmitting an audio signal from the DSP to the RAP, mixing the audio signal and the anti-noise signal in the RAP, and generating an expected output signal when generating the anti-noise signal. The method further includes reducing the cancellation of the audio signal due to the anti-noise signal by setting the reference point.

  Example 20 includes the method of Examples 13-19, wherein the method supports adjusting a DAC amplifier based on the anti-noise signal level by transferring the anti-noise signal to a digital-to-analog converter (DAC) amplifier controller. It further includes.

  The foregoing embodiments of the disclosed subject matter have many advantages that have been described or will be apparent to those skilled in the art. However, not all disclosed devices, systems, or methods need to include all of these advantages or features.

  Furthermore, written statements refer to particular features. It is to be understood that the disclosed specification includes all possible combinations of these specific features. Certain features disclosed in a particular aspect or embodiment can be applied to other aspects and embodiments to the extent possible.

  Also, in this application, when referring to a method having two or more predetermined steps or operations, the predetermined steps or operations may be performed in any order or concurrently, unless context precludes that possibility. .

  While certain examples of the disclosure have been illustrated and described for purposes of illustration, it should be understood that various modifications can be made without departing from the spirit and scope of the disclosure. Accordingly, the present disclosure is limited only by the appended claims.

Claims (20)

A sound processing network,
A digital signal processor (DSP) operating at a first frequency;
The DSP is:
Receiving a noise signal from at least one microphone; and generating a noise filter based on the noise signal;
A real-time sound processor (RAP) operating at a second frequency higher than the first frequency;
The RAP is
Receiving the noise signal from the microphone;
An acoustic processing network that receives the noise filter from the DSP and generates an anti-noise signal for use in active noise cancellation (ANC) based on the noise signal and the noise filter. The RAP is
An adjustable amplifier for amplifying the anti-noise signal;
The audio processing network of claim 1, further comprising: a compressor circuit for controlling disturbance of the anti-noise signal by controlling the adjustable amplifier. The RAP further includes a compression status register for storing a compression status,
The sound processing network of claim 2, wherein the compressor circuit further controls the adjustable amplifier based on the compression state.   The sound processing network according to claim 3, wherein the compression state includes a peak signal estimate, an instantaneous gain, a target gain, an attack parameter, a release parameter, an attenuation parameter, a maintenance parameter, or a combination thereof.   4. The sound processing network of claim 3, wherein the compression state comprises a root mean square (RMS) of the anti-noise signal. The DSP further receives a current compression state from the RAP,
3. Support control of the adjustable amplifier by determining a new compression state based on the noise signal and the current compression state, and forwarding the new compression state to the RAP. A sound processing network according to claim 1.   The acoustic processing network of claim 1, wherein the RAP implements the noise filter from the DSP and includes one or more programmable biquad filters to generate the anti-noise signal. The biquad filter amplifies a portion of the sample of the anti-noise signal using one or more poles and attenuates a portion of the sample of the anti-noise signal using one or more zeros. Storing the quantization of the samples of the anti-noise signal using a filter register;
The acoustic processing network of claim 7, wherein the biquad filter amplifies the sample before quantizing the sample, and then attenuates the sample. The microphone is a feedforward microphone;
The RAP further generates an enhanced predetermined frequency band by applying an environment recognition filter to enhance the predetermined frequency band of the noise signal when generating the anti-noise signal;
The sound processing network according to claim 1, wherein the anti-noise signal including the enhanced predetermined frequency band is transferred to a speaker and output to a user.   The acoustic processing network of claim 9, wherein a delay between receiving a noise signal sample from the microphone and transferring a corresponding anti-noise signal sample to the speaker is less than 100 microseconds. The DSP further generating an audio signal based on the audio input, and generating an expected output signal based on the audio input and the frequency response of the audio processing network;
The RAP further receives the audio signal from the DSP,
Mixing the audio signal and the anti-noise signal, and reducing the cancellation of the audio signal by the anti-noise signal by setting the expected output signal as a reference point when generating the anti-noise signal The sound processing network according to claim 1, wherein:   The RAP of claim 1, wherein the RAP is further configured to support adjustment of a DAC amplifier based on an anti-noise signal level by transferring the anti-noise signal to a digital-to-analog converter (DAC) amplifier controller. Sound processing network. The method
Receiving a noise signal at a digital signal processor (DSP) operating at a first frequency, wherein the noise signal is received from at least one microphone;
Generating a noise filter in the DSP based on the noise signal;
Transmitting the noise filter from the DSP to a real-time sound processor (RAP) operating at a second frequency higher than the first frequency;
Receiving the noise signal from the microphone at the RAP;
Generating an anti-noise signal for use in active noise cancellation (ANC) at the RAP based on the noise signal and the noise filter. Adjusting the anti-noise signal by controlling an adjustable amplifier using the current compression state at the RAP;
Transmitting the current compression state from the RAP to the DSP;
Determining a new compression state in the DSP based on the noise signal and the current compression state;
14. The method of claim 13, further comprising: transmitting the new compression state from the DSP to the RAP to support control of the adjustable amplifier.   The method of claim 14, wherein the compression state comprises a peak signal estimate, an instantaneous gain, a target gain, a root mean square (RMS), or a combination thereof, of the anti-noise signal.   14. The method of claim 13, wherein the anti-noise signal is generated at the RAP by configuring one or more programmable biquad filters to implement the noise filter from the DSP.   17. The method of claim 16, wherein the biquad filter amplifies the samples of the anti-noise signal before quantizing the samples of the anti-noise signal and then attenuating the samples of the anti-noise filter. Generating an enhanced predetermined frequency band by applying an environment recognition filter in the RAP to enhance a predetermined frequency band of the noise signal when generating the anti-noise signal;
14. The method of claim 13, further comprising: transferring the anti-noise signal including the enhanced predetermined frequency band to a speaker for output to a user. Generating an audio signal with the DSP based on the audio input;
Generating an expected output signal at the DSP based on the speech input and the frequency response of the sound processing network;
Transmitting the audio signal from the DSP to the RAP;
Mixing the audio signal and the anti-noise signal at the RAP;
14. The method of claim 13, further comprising: when generating the anti-noise signal, reducing the cancellation of the audio signal by the anti-noise signal by setting the expected output signal as a reference point. .   14. The method of claim 13, further comprising supporting adjustment of a DAC amplifier based on an anti-noise signal level by transferring the anti-noise signal to a digital-to-analog converter (DAC) amplifier controller.

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